1. Field of the Invention
The present invention relates to a signal processing device and a signal processing method that automatically control a level of an input signal.
2. Related Art
In digital video cameras and digital still cameras, a device (hereinafter, referred to as a signal processing device) for converting input analog audio signals into digital signals, controlling tones and the like is provided at an input path for recording of motion picture sounds.
For example, a signal processing device disclosed in Japanese Patent Application Laid Open (JP-A) No. 2009-273045 is known. FIG. 3 is a view showing an exemplary configuration of a signal processor 10 described in JP-A No. 2009-273045. The signal processor 10 includes an input terminal 12, an analog signal processing block 14, a digital signal processing block 16, an auto level controller (ALC) 18, a level difference detector 20, a correction gain device 22, and an output terminal 24.
An analog audio signal input to the input terminal 12 through an unillustrated microphone is input to the analog signal processing block 14. The analog signal processing block 14 includes a programmable gain amplifier (PGA) 32 serving as an input amplifier and an analog-to-digital converter (ADC) 34. The analog audio signal input to the analog signal processing block 14 is first amplified by the PGA 32 and then converted into a digital signal by the ADC 34, and the digital signal is input to the subsequent digital signal processing block 16.
The PGA 32 is an externally programmable amplifier, the gain of which can be set to an arbitrary value. The gain of the PGA 32 is set to a value in accordance with a control signal given from the ALC 18, and the PGA 32 amplifies the input analog audio signal with the gain thus set.
The digital audio signal into which the analog audio signal is converted by the ADC 34 is input to the digital signal processing block 16. The digital signal processing block 16 includes a decimation filter 36, a DC-cut high pass filter (hereinafter, referred to as a DC-cut HPF) 38, and a tone control filtering section 40.
The decimation filter 36 performs decimation of the digital audio signal input from the ADC 34 and outputs the digital audio signal with a reduced sampling rate to the DC-cut HPF 38. In response to the receiving of the decimated digital audio signal, the DC-cut HPF 38 cuts out superfluous DC components generated by digital conversion in the ADC 34.
The ALC 18 is provided subsequent to the DC-cut HPF 38. The ALC 18 detects a level (magnitude of a sound, i.e., amplitude thereof) of the digital audio signal input from the DC-cut HPF 38 and sets the gain of the PGA 32 on the basis of the detected level so that the analog audio signal input to the PGA 32 can be amplified to a constant level. The input audio level is thereby automatically controlled to a constant level. The ALC 18 is a control circuit for detecting the level of the digital audio signal and outputting a control signal to the PGA 32 and outputs the input digital audio signal to the subsequent tone control filtering section 40 and the level difference detector 20 without performing processing on the input digital audio signal.
The tone control filtering section 40 performs filtering for tone control. The tone control filtering section 40 includes a wind noise removal high pass filter (HPF) 42 and a notch filter 44. The wind noise removal HPF 42 removes a low-frequency wind noise component which is input through the microphone. This filter is used with a predetermined cut-off frequency (e.g., about 100 to 200 Hz). After removing the wind noise, the notch filter 44 removes noise of a specific frequency which occurs depending on the equipment in which this device is included.
The digital audio signal, the tone of which is controlled by the tone control filtering section 40 is input to the level difference detector 20 and the correction gain device 22 which are provided at the stage subsequent thereto.
The digital audio signal before being filtered by the tone control filtering section 40 and the digital audio signal after being filtered by the tone control filtering section 40 are input to the level difference detector 20. The level difference detector 20 detects the difference in the level between the two input digital audio signals and outputs the detected difference to the correction gain device 22 as a correction gain value.
The tone-controlled digital audio signal from the notch filter 44 and the correction gain value from the level difference detector 20 are input to the correction gain device 22. The correction gain device 22 amplifies the input digital audio signal in accordance with the input correction gain value so that the digital audio signal has a predetermined constant level. After the level, correction by the correction gain device 22, the resultant digital audio signal is output as recorded data from the output terminal 24.
The signal processor shown in FIG. 3 can always obtain an output of constant level and prevent a reduction in sound volume due to cut-off of low-frequency sound or the like since the level attenuated by the tone control filtering section 40 (the wind noise removal HPF 42 and the notch filter 44) is compensated for by the correction gain device 22.
However, since the level control is performed only in the analog area having a wide dynamic range (having a large difference between a large sound and a small sound), in some cases, control for obtaining a desired sound may be difficult depending on the type of input sound signal. Specific discussion thereon will be provided below.
Supposing there is an occurrence of very loud sound, it is preferable to set a target level of sound in the PGA 32 to a low level (e.g., −4 dBFS or the like) to some extent in advance, and also to set a decay rate (a rate of gradually increasing the gain in accordance with a decrease in the level of the audio signal after the gain is reduced) to be low to some extent. This is because if the target level is set relatively low, since the gain is controlled so that the level of the input analog audio signal may start decreasing at the point in time when the level of the input analog audio signal exceeds the target level, it is possible to suppress frequent occurrence of saturation due to belated gain reduction at the start of the very loud sound. Further, by setting the decay rate to be low, the gain would not easily increase, thereby reducing the frequency of saturation.
However, in any of above-mentioned case, the sound volume which is obtained as a result becomes relatively low on the whole.
Accordingly, in a device having the conventional configuration described above, it is difficult in some cases to increase the sound volume while suppressing saturation.
Further, the signal processor shown in FIG. 3 is configured so as to perform gain control based on the difference in envelopes before and after the tone control filtering section 40. Therefore, even if an input audio level is at a substantially silent level, a minute difference may occur due to the effect of a quantization error caused by the filtering operation in the tone control filtering section 40, and by detecting this minute difference, the signal processor may increase the gain too much and raise a floor noise level excessively.